int WebRtcNetEQ_RecInInternal(MCUInst_t *MCU_inst, RTPPacket_t *RTPpacketInput, WebRtc_UWord32 uw32_timeRec) { RTPPacket_t RTPpacket[2]; int i_k; int i_ok = 0, i_No_Of_Payloads = 1; WebRtc_Word16 flushed = 0; WebRtc_Word16 codecPos; WebRtc_UWord32 diffTS, uw32_tmp; int curr_Codec; WebRtc_Word16 isREDPayload = 0; WebRtc_Word32 temp_bufsize = MCU_inst->PacketBuffer_inst.numPacketsInBuffer; #ifdef NETEQ_RED_CODEC RTPPacket_t* RTPpacketPtr[2]; /* Support for redundancy up to 2 payloads */ RTPpacketPtr[0] = &RTPpacket[0]; RTPpacketPtr[1] = &RTPpacket[1]; #endif /* * Copy from input RTP packet to local copy * (mainly to enable multiple payloads using RED) */ WEBRTC_SPL_MEMCPY_W8(&RTPpacket[0], RTPpacketInput, sizeof(RTPPacket_t)); /* Reinitialize NetEq if it's needed (changed SSRC or first call) */ if ((RTPpacket[0].ssrc != MCU_inst->ssrc) || (MCU_inst->first_packet == 1)) { WebRtcNetEQ_RTCPInit(&MCU_inst->RTCP_inst, RTPpacket[0].seqNumber); MCU_inst->first_packet = 0; /* Flush the buffer */ WebRtcNetEQ_PacketBufferFlush(&MCU_inst->PacketBuffer_inst); /* Store new SSRC */ MCU_inst->ssrc = RTPpacket[0].ssrc; /* Update codecs */ MCU_inst->timeStamp = RTPpacket[0].timeStamp; MCU_inst->current_Payload = RTPpacket[0].payloadType; /*Set MCU to update codec on next SignalMCU call */ MCU_inst->new_codec = 1; /* Reset timestamp scaling */ MCU_inst->TSscalingInitialized = 0; } /* Call RTCP statistics */ i_ok |= WebRtcNetEQ_RTCPUpdate(&(MCU_inst->RTCP_inst), RTPpacket[0].seqNumber, RTPpacket[0].timeStamp, uw32_timeRec); /* If Redundancy is supported and this is the redundancy payload, separate the payloads */ #ifdef NETEQ_RED_CODEC if (RTPpacket[0].payloadType == WebRtcNetEQ_DbGetPayload(&MCU_inst->codec_DB_inst, kDecoderRED)) { /* Split the payload into a main and a redundancy payloads */ i_ok = WebRtcNetEQ_RedundancySplit(RTPpacketPtr, 2, &i_No_Of_Payloads); if (i_ok < 0) { /* error returned */ return i_ok; } /* * Only accept a few redundancies of the same type as the main data, * AVT events and CNG. */ if ((i_No_Of_Payloads > 1) && (RTPpacket[0].payloadType != RTPpacket[1].payloadType) && (RTPpacket[0].payloadType != WebRtcNetEQ_DbGetPayload(&MCU_inst->codec_DB_inst, kDecoderAVT)) && (RTPpacket[1].payloadType != WebRtcNetEQ_DbGetPayload( &MCU_inst->codec_DB_inst, kDecoderAVT)) && (!WebRtcNetEQ_DbIsCNGPayload( &MCU_inst->codec_DB_inst, RTPpacket[0].payloadType)) && (!WebRtcNetEQ_DbIsCNGPayload(&MCU_inst->codec_DB_inst, RTPpacket[1].payloadType))) { i_No_Of_Payloads = 1; } isREDPayload = 1; } #endif /* loop over the number of payloads */ for (i_k = 0; i_k < i_No_Of_Payloads; i_k++) { if (isREDPayload == 1) { RTPpacket[i_k].rcuPlCntr = i_k; } else { RTPpacket[i_k].rcuPlCntr = 0; } /* Force update of SplitInfo if it's iLBC because of potential change between 20/30ms */ if (RTPpacket[i_k].payloadType == WebRtcNetEQ_DbGetPayload(&MCU_inst->codec_DB_inst, kDecoderILBC)) { i_ok = WebRtcNetEQ_DbGetSplitInfo( &MCU_inst->PayloadSplit_inst, (enum WebRtcNetEQDecoder) WebRtcNetEQ_DbGetCodec(&MCU_inst->codec_DB_inst, RTPpacket[i_k].payloadType), RTPpacket[i_k].payloadLen); if (i_ok < 0) { /* error returned */ return i_ok; } } /* Get information about timestamp scaling for this payload type */ i_ok = WebRtcNetEQ_GetTimestampScaling(MCU_inst, RTPpacket[i_k].payloadType); if (i_ok < 0) { /* error returned */ return i_ok; } if (MCU_inst->TSscalingInitialized == 0 && MCU_inst->scalingFactor != kTSnoScaling) { /* Must initialize scaling with current timestamps */ MCU_inst->externalTS = RTPpacket[i_k].timeStamp; MCU_inst->internalTS = RTPpacket[i_k].timeStamp; MCU_inst->TSscalingInitialized = 1; } /* Adjust timestamp if timestamp scaling is needed (e.g. SILK or G.722) */ if (MCU_inst->TSscalingInitialized == 1) { WebRtc_UWord32 newTS = WebRtcNetEQ_ScaleTimestampExternalToInternal(MCU_inst, RTPpacket[i_k].timeStamp); /* save the incoming timestamp for next time */ MCU_inst->externalTS = RTPpacket[i_k].timeStamp; /* add the scaled difference to last scaled timestamp and save ... */ MCU_inst->internalTS = newTS; RTPpacket[i_k].timeStamp = newTS; } /* Is this a DTMF packet?*/ if (RTPpacket[i_k].payloadType == WebRtcNetEQ_DbGetPayload(&MCU_inst->codec_DB_inst, kDecoderAVT)) { #ifdef NETEQ_ATEVENT_DECODE if (MCU_inst->AVT_PlayoutOn) { i_ok = WebRtcNetEQ_DtmfInsertEvent(&MCU_inst->DTMF_inst, RTPpacket[i_k].payload, RTPpacket[i_k].payloadLen, RTPpacket[i_k].timeStamp); if (i_ok != 0) { return i_ok; } } #endif #ifdef NETEQ_STEREO if (MCU_inst->usingStereo == 0) { /* do not set this for DTMF packets when using stereo mode */ MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF = 1; } #else MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF = 1; #endif } else if (WebRtcNetEQ_DbIsCNGPayload(&MCU_inst->codec_DB_inst, RTPpacket[i_k].payloadType)) { /* Is this a CNG packet? how should we handle this?*/ #ifdef NETEQ_CNG_CODEC /* Get CNG sample rate */ WebRtc_UWord16 fsCng = WebRtcNetEQ_DbGetSampleRate(&MCU_inst->codec_DB_inst, RTPpacket[i_k].payloadType); if ((fsCng != MCU_inst->fs) && (fsCng > 8000)) { /* * We have received CNG with a different sample rate from what we are using * now (must be > 8000, since we may use only one CNG type (default) for all * frequencies). Flush buffer and signal new codec. */ WebRtcNetEQ_PacketBufferFlush(&MCU_inst->PacketBuffer_inst); MCU_inst->new_codec = 1; MCU_inst->current_Codec = -1; } i_ok = WebRtcNetEQ_PacketBufferInsert(&MCU_inst->PacketBuffer_inst, &RTPpacket[i_k], &flushed); if (i_ok < 0) { return RECIN_CNG_ERROR; } MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF = 1; #else /* NETEQ_CNG_CODEC not defined */ return RECIN_UNKNOWNPAYLOAD; #endif /* NETEQ_CNG_CODEC */ } else { /* Reinitialize the splitting if the payload and/or the payload length has changed */ curr_Codec = WebRtcNetEQ_DbGetCodec(&MCU_inst->codec_DB_inst, RTPpacket[i_k].payloadType); if (curr_Codec != MCU_inst->current_Codec) { if (curr_Codec < 0) { return RECIN_UNKNOWNPAYLOAD; } MCU_inst->current_Codec = curr_Codec; MCU_inst->current_Payload = RTPpacket[i_k].payloadType; i_ok = WebRtcNetEQ_DbGetSplitInfo(&MCU_inst->PayloadSplit_inst, (enum WebRtcNetEQDecoder) MCU_inst->current_Codec, RTPpacket[i_k].payloadLen); if (i_ok < 0) { /* error returned */ return i_ok; } WebRtcNetEQ_PacketBufferFlush(&MCU_inst->PacketBuffer_inst); MCU_inst->new_codec = 1; } /* update post-call statistics */ if (MCU_inst->new_codec != 1) /* not if in codec change */ { diffTS = RTPpacket[i_k].timeStamp - MCU_inst->timeStamp; /* waiting time */ if (diffTS < 0x0FFFFFFF) /* guard against re-ordering */ { /* waiting time in ms */ diffTS = WEBRTC_SPL_UDIV(diffTS, WEBRTC_SPL_UDIV((WebRtc_UWord32) MCU_inst->fs, 1000) ); if (diffTS < MCU_inst->statInst.minPacketDelayMs) { /* new all-time low */ MCU_inst->statInst.minPacketDelayMs = diffTS; } if (diffTS > MCU_inst->statInst.maxPacketDelayMs) { /* new all-time high */ MCU_inst->statInst.maxPacketDelayMs = diffTS; } /* Update avg waiting time: * avgPacketDelayMs = * (avgPacketCount * avgPacketDelayMs + diffTS)/(avgPacketCount+1) * with proper rounding. */ uw32_tmp = WEBRTC_SPL_UMUL((WebRtc_UWord32) MCU_inst->statInst.avgPacketCount, (WebRtc_UWord32) MCU_inst->statInst.avgPacketDelayMs); uw32_tmp = WEBRTC_SPL_ADD_SAT_W32(uw32_tmp, (diffTS + (MCU_inst->statInst.avgPacketCount>>1))); uw32_tmp = WebRtcSpl_DivU32U16(uw32_tmp, (WebRtc_UWord16) (MCU_inst->statInst.avgPacketCount + 1)); MCU_inst->statInst.avgPacketDelayMs = (WebRtc_UWord16) WEBRTC_SPL_MIN(uw32_tmp, (WebRtc_UWord32) 65535); /* increase counter, but not to more than 65534 */ if (MCU_inst->statInst.avgPacketCount < (0xFFFF - 1)) { MCU_inst->statInst.avgPacketCount++; } } } /* Parse the payload and insert it into the buffer */ i_ok = WebRtcNetEQ_SplitAndInsertPayload(&RTPpacket[i_k], &MCU_inst->PacketBuffer_inst, &MCU_inst->PayloadSplit_inst, &flushed); if (i_ok < 0) { return i_ok; } if (MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF != 0) { /* first normal packet after CNG or DTMF */ MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF = -1; } }
int WebRtcNetEQ_RecInInternal(MCUInst_t *MCU_inst, RTPPacket_t *RTPpacketInput, WebRtc_UWord32 uw32_timeRec) { RTPPacket_t RTPpacket[2]; int i_k; int i_ok = 0, i_No_Of_Payloads = 1; WebRtc_Word16 flushed = 0; WebRtc_Word16 codecPos; int curr_Codec; WebRtc_Word16 isREDPayload = 0; WebRtc_Word32 temp_bufsize = MCU_inst->PacketBuffer_inst.numPacketsInBuffer; #ifdef NETEQ_RED_CODEC RTPPacket_t* RTPpacketPtr[2]; /* Support for redundancy up to 2 payloads */ RTPpacketPtr[0] = &RTPpacket[0]; RTPpacketPtr[1] = &RTPpacket[1]; #endif /* * Copy from input RTP packet to local copy * (mainly to enable multiple payloads using RED) */ WEBRTC_SPL_MEMCPY_W8(&RTPpacket[0], RTPpacketInput, sizeof(RTPPacket_t)); /* Reinitialize NetEq if it's needed (changed SSRC or first call) */ if ((RTPpacket[0].ssrc != MCU_inst->ssrc) || (MCU_inst->first_packet == 1)) { WebRtcNetEQ_RTCPInit(&MCU_inst->RTCP_inst, RTPpacket[0].seqNumber); MCU_inst->first_packet = 0; /* Flush the buffer */ WebRtcNetEQ_PacketBufferFlush(&MCU_inst->PacketBuffer_inst); /* Store new SSRC */ MCU_inst->ssrc = RTPpacket[0].ssrc; /* Update codecs */ MCU_inst->timeStamp = RTPpacket[0].timeStamp; MCU_inst->current_Payload = RTPpacket[0].payloadType; /*Set MCU to update codec on next SignalMCU call */ MCU_inst->new_codec = 1; /* Reset timestamp scaling */ MCU_inst->TSscalingInitialized = 0; } /* Call RTCP statistics */ i_ok |= WebRtcNetEQ_RTCPUpdate(&(MCU_inst->RTCP_inst), RTPpacket[0].seqNumber, RTPpacket[0].timeStamp, uw32_timeRec); /* If Redundancy is supported and this is the redundancy payload, separate the payloads */ #ifdef NETEQ_RED_CODEC if (RTPpacket[0].payloadType == WebRtcNetEQ_DbGetPayload(&MCU_inst->codec_DB_inst, kDecoderRED)) { /* Split the payload into a main and a redundancy payloads */ i_ok = WebRtcNetEQ_RedundancySplit(RTPpacketPtr, 2, &i_No_Of_Payloads); if (i_ok < 0) { /* error returned */ return i_ok; } /* * Only accept a few redundancies of the same type as the main data, * AVT events and CNG. */ if ((i_No_Of_Payloads > 1) && (RTPpacket[0].payloadType != RTPpacket[1].payloadType) && (RTPpacket[0].payloadType != WebRtcNetEQ_DbGetPayload(&MCU_inst->codec_DB_inst, kDecoderAVT)) && (RTPpacket[1].payloadType != WebRtcNetEQ_DbGetPayload( &MCU_inst->codec_DB_inst, kDecoderAVT)) && (!WebRtcNetEQ_DbIsCNGPayload( &MCU_inst->codec_DB_inst, RTPpacket[0].payloadType)) && (!WebRtcNetEQ_DbIsCNGPayload(&MCU_inst->codec_DB_inst, RTPpacket[1].payloadType))) { i_No_Of_Payloads = 1; } isREDPayload = 1; } #endif /* loop over the number of payloads */ for (i_k = 0; i_k < i_No_Of_Payloads; i_k++) { if (isREDPayload == 1) { RTPpacket[i_k].rcuPlCntr = i_k; } else { RTPpacket[i_k].rcuPlCntr = 0; } /* Force update of SplitInfo if it's iLBC because of potential change between 20/30ms */ if (RTPpacket[i_k].payloadType == WebRtcNetEQ_DbGetPayload(&MCU_inst->codec_DB_inst, kDecoderILBC)) { i_ok = WebRtcNetEQ_DbGetSplitInfo( &MCU_inst->PayloadSplit_inst, (enum WebRtcNetEQDecoder) WebRtcNetEQ_DbGetCodec(&MCU_inst->codec_DB_inst, RTPpacket[i_k].payloadType), RTPpacket[i_k].payloadLen); if (i_ok < 0) { /* error returned */ return i_ok; } } /* Get information about timestamp scaling for this payload type */ i_ok = WebRtcNetEQ_GetTimestampScaling(MCU_inst, RTPpacket[i_k].payloadType); if (i_ok < 0) { /* error returned */ return i_ok; } if (MCU_inst->TSscalingInitialized == 0 && MCU_inst->scalingFactor != kTSnoScaling) { /* Must initialize scaling with current timestamps */ MCU_inst->externalTS = RTPpacket[i_k].timeStamp; MCU_inst->internalTS = RTPpacket[i_k].timeStamp; MCU_inst->TSscalingInitialized = 1; } /* Adjust timestamp if timestamp scaling is needed (e.g. SILK or G.722) */ if (MCU_inst->TSscalingInitialized == 1) { WebRtc_UWord32 newTS = WebRtcNetEQ_ScaleTimestampExternalToInternal(MCU_inst, RTPpacket[i_k].timeStamp); /* save the incoming timestamp for next time */ MCU_inst->externalTS = RTPpacket[i_k].timeStamp; /* add the scaled difference to last scaled timestamp and save ... */ MCU_inst->internalTS = newTS; RTPpacket[i_k].timeStamp = newTS; } /* Is this a DTMF packet?*/ if (RTPpacket[i_k].payloadType == WebRtcNetEQ_DbGetPayload(&MCU_inst->codec_DB_inst, kDecoderAVT)) { #ifdef NETEQ_ATEVENT_DECODE if (MCU_inst->AVT_PlayoutOn) { i_ok = WebRtcNetEQ_DtmfInsertEvent(&MCU_inst->DTMF_inst, RTPpacket[i_k].payload, RTPpacket[i_k].payloadLen, RTPpacket[i_k].timeStamp); if (i_ok != 0) { return i_ok; } } #endif #ifdef NETEQ_STEREO if (MCU_inst->usingStereo == 0) { /* do not set this for DTMF packets when using stereo mode */ MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF = 1; } #else MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF = 1; #endif } else if (WebRtcNetEQ_DbIsCNGPayload(&MCU_inst->codec_DB_inst, RTPpacket[i_k].payloadType)) { /* Is this a CNG packet? how should we handle this?*/ #ifdef NETEQ_CNG_CODEC /* Get CNG sample rate */ WebRtc_UWord16 fsCng = WebRtcNetEQ_DbGetSampleRate(&MCU_inst->codec_DB_inst, RTPpacket[i_k].payloadType); /* Force sampling frequency to 32000 Hz CNG 48000 Hz. */ /* TODO(tlegrand): remove limitation once ACM has full 48 kHz * support. */ if (fsCng > 32000) { fsCng = 32000; } if ((fsCng != MCU_inst->fs) && (fsCng > 8000)) { /* * We have received CNG with a different sample rate from what we are using * now (must be > 8000, since we may use only one CNG type (default) for all * frequencies). Flush buffer and signal new codec. */ WebRtcNetEQ_PacketBufferFlush(&MCU_inst->PacketBuffer_inst); MCU_inst->new_codec = 1; MCU_inst->current_Codec = -1; } i_ok = WebRtcNetEQ_PacketBufferInsert(&MCU_inst->PacketBuffer_inst, &RTPpacket[i_k], &flushed); if (i_ok < 0) { return RECIN_CNG_ERROR; } MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF = 1; #else /* NETEQ_CNG_CODEC not defined */ return RECIN_UNKNOWNPAYLOAD; #endif /* NETEQ_CNG_CODEC */ } else { /* Reinitialize the splitting if the payload and/or the payload length has changed */ curr_Codec = WebRtcNetEQ_DbGetCodec(&MCU_inst->codec_DB_inst, RTPpacket[i_k].payloadType); if (curr_Codec != MCU_inst->current_Codec) { if (curr_Codec < 0) { return RECIN_UNKNOWNPAYLOAD; } MCU_inst->current_Codec = curr_Codec; MCU_inst->current_Payload = RTPpacket[i_k].payloadType; i_ok = WebRtcNetEQ_DbGetSplitInfo(&MCU_inst->PayloadSplit_inst, (enum WebRtcNetEQDecoder) MCU_inst->current_Codec, RTPpacket[i_k].payloadLen); if (i_ok < 0) { /* error returned */ return i_ok; } WebRtcNetEQ_PacketBufferFlush(&MCU_inst->PacketBuffer_inst); MCU_inst->new_codec = 1; } /* Parse the payload and insert it into the buffer */ i_ok = WebRtcNetEQ_SplitAndInsertPayload(&RTPpacket[i_k], &MCU_inst->PacketBuffer_inst, &MCU_inst->PayloadSplit_inst, &flushed); if (i_ok < 0) { return i_ok; } if (MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF != 0) { /* first normal packet after CNG or DTMF */ MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF = -1; } } /* Reset DSP timestamp etc. if packet buffer flushed */ if (flushed) { MCU_inst->new_codec = 1; } } /* * Update Bandwidth Estimate * Only send the main payload to BWE */ if ((curr_Codec = WebRtcNetEQ_DbGetCodec(&MCU_inst->codec_DB_inst, RTPpacket[0].payloadType)) >= 0) { codecPos = MCU_inst->codec_DB_inst.position[curr_Codec]; if (MCU_inst->codec_DB_inst.funcUpdBWEst[codecPos] != NULL) /* codec has BWE function */ { if (RTPpacket[0].starts_byte1) /* check for shifted byte alignment */ { /* re-align to 16-bit alignment */ for (i_k = 0; i_k < RTPpacket[0].payloadLen; i_k++) { WEBRTC_SPL_SET_BYTE(RTPpacket[0].payload, WEBRTC_SPL_GET_BYTE(RTPpacket[0].payload, i_k+1), i_k); } RTPpacket[0].starts_byte1 = 0; } MCU_inst->codec_DB_inst.funcUpdBWEst[codecPos]( MCU_inst->codec_DB_inst.codec_state[codecPos], (G_CONST WebRtc_UWord16 *) RTPpacket[0].payload, (WebRtc_Word32) RTPpacket[0].payloadLen, RTPpacket[0].seqNumber, (WebRtc_UWord32) RTPpacket[0].timeStamp, (WebRtc_UWord32) uw32_timeRec); } } if (MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF == 0) { /* Calculate the total speech length carried in each packet */ temp_bufsize = MCU_inst->PacketBuffer_inst.numPacketsInBuffer - temp_bufsize; temp_bufsize *= MCU_inst->PacketBuffer_inst.packSizeSamples; if ((temp_bufsize > 0) && (MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF == 0) && (temp_bufsize != MCU_inst->BufferStat_inst.Automode_inst.packetSpeechLenSamp)) { /* Change the auto-mode parameters if packet length has changed */ WebRtcNetEQ_SetPacketSpeechLen(&(MCU_inst->BufferStat_inst.Automode_inst), (WebRtc_Word16) temp_bufsize, MCU_inst->fs); } /* update statistics */ if ((WebRtc_Word32) (RTPpacket[0].timeStamp - MCU_inst->timeStamp) >= 0 && !MCU_inst->new_codec) { /* * Only update statistics if incoming packet is not older than last played out * packet, and if new codec flag is not set. */ WebRtcNetEQ_UpdateIatStatistics(&MCU_inst->BufferStat_inst.Automode_inst, MCU_inst->PacketBuffer_inst.maxInsertPositions, RTPpacket[0].seqNumber, RTPpacket[0].timeStamp, MCU_inst->fs, WebRtcNetEQ_DbIsMDCodec((enum WebRtcNetEQDecoder) MCU_inst->current_Codec), (MCU_inst->NetEqPlayoutMode == kPlayoutStreaming)); } } else if (MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF == -1) { /* * This is first "normal" packet after CNG or DTMF. * Reset packet time counter and measure time until next packet, * but don't update statistics. */ MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF = 0; MCU_inst->BufferStat_inst.Automode_inst.packetIatCountSamp = 0; } return 0; }